process_input_packet() contains two non-interacting pieces of nontrivial
size and complexity - decoding and streamcopy. Separating them makes the
code easier to read.
New placement requires fewer explicit conditions and is easier to
understand.
The logic should be exactly equivalent, since this is the only place
where eof_reached is set for decoding.
Passing ist=NULL is currently used to identify stream types that do not
decode into AVFrames, i.e. subtitles. That is highly non-obvious -
always pass a non-NULL InputStream and just check the type explicitly.
It tracks whether the decoder for this stream ever produced any frames
and its only use is for checking whether a filter input ever received a
frame - those that did not are prioritized by the scheduler.
This is awkward and unnecessarily complicated - checking whether the
filtergraph input format is valid works just as well and does not
require maintaining an extra variable.
It does no initialization anymore, except for setting
transcode_init_done - the bulk of the function is printing the
input/output maps. It also cannot fail anymore, so remove the useless
return value.
Export the corresponding flag in InputFile instead. This will allow
making the demuxer AVFormatContext private in future commits, similarly
to what was previously done for muxers.
There is no point in having a per-stream wallclock start time, since
they are all computed at the same instant. Keep a per-file start time
instead, initialized when the demuxer thread starts.
That is a more appropriate place for this code and will allow hiding
more of InputStream.
The value of repeat_pict extracted from libavformat internal parser no
longer needs to be trasmitted outside of the demuxing thread.
Move readrate handling to the demuxer thread. This has to be done in the
same commit, since it reads InputStream.dts,nb_packets, which are now
set in the demuxer thread.
This way computing it and using it for streamcopy does not need to
happen in sync. Will be useful in following commits, where updating
InputStream.dts will be moved to the demuxing thread.
This code runs post-demuxing and is not synchronized with the decoder
output (which may be delayed with respect to its input by arbitrary and
unknowable amounts), so accessing any decoder properties is incorrect.
Move them to a separate function called right after timestamp
discontinuity processing. This is now possible, since these values have
no interaction with decoding anymore.
When an input stream terminates and no frames were successfully decoded,
filtering code will currently configure the filtergraph using demuxer
stream parameters. Use decoder parameters instead, which should be more
reliable. Also, initialize them immediately when an input stream is
bound to a filtergraph input, so that these parameters are always
available (if at all) and filtering code does not need to reach into the
decoder at some arbitrary later point.
These two functions are a part of a single logical action - determining
which, if any, output stream needs to be processed next. Keeping them
separate is a historical artifact that obscures what is actually being
done.
Currently those are set in different ways depending on whether the
stream is decoded or not, using some values from the decoder if it is.
This is wrong, because there may be arbitrary amount of delay between
input packets and output frames (depending e.g. on the thread count when
frame threading is used).
Always use the path that was previously used only for streamcopy. This
should not cause any issues, because these values are now used only for
streamcopy and discontinuity handling.
This change will allow to decouple discontinuity processing from
decoding and move it to ffmpeg_demux. It also makes the code simpler.
Changes output in fate-cover-art-aiff-id3v2-remux and
fate-cover-art-mp3-id3v2-remux, where attached pictures are now written
in the correct order. This happens because InputStream.dts is no longer
reset to AV_NOPTS_VALUE after decoding, so streamcopy actually sees
valid dts values.
This was added in 380db56928 as a
temporary crutch that is not needed anymore. The only case where this
code can be triggered is the very first frame, for which InputStream.pts
is always equal to 0.
Stop using InputStream.dts for generating missing timestamps for decoded
frames, because it contains pre-decoding timestamps and there may be
arbitrary amount of delay between input packets and output frames (e.g.
dependent on the thread count when frame threading is used). It is also
in AV_TIME_BASE (i.e. microseconds), which may introduce unnecessary
rounding issues.
New code maintains a timebase that is the inverse of the LCM of all the
samplerates seen so far, and thus can accurately represent every audio
sample. This timebase is used to generate missing timestamps after
decoding.
Changes the result of the following FATE tests
* pcm_dvd-16-5.1-96000
* lavf-smjpeg
* adpcm-ima-smjpeg
In all of these the timestamps now better correspond to actual frame
durations.
If input packets have timestamps, they will be propagated to output
frames by the decoder, so at best this block does not do anything.
There can also be an arbitrary amount of delay between packets sent to
the decoder and decoded frames (e.g. due to decoder's intrinsic delay or
frame threading), so deriving any timestamps from packet properties is
wrong.
ts_discontinuity_detect() is applied right after demuxing, while
InputStream.pts is a post-decoding timestamp, which may be delayed with
respect to demuxing by an arbitrary amount (e.g. depending on the thread
count when frame threading is used).
Anton Khirnov
James Almer